Added following snippits about Asterisk

* A Basic Asterisk server for internal calling with IVR, curl triggers and message playback
* A Asterisk server to act as a bridge between a Bluetooth phone (as in/out bound route)
* A Node-Red Subflow to create a call between extensions using ARI

Linux/Config-Examples/Asterisk-BT_bridge/ReadMe.md

@@ -0,0 +1,27 @@
+# Asterisk sever BT bridge
+Go [Here](../../Docs/Setup-Asterisk-Server.md) to read about how to set it up
+
+
+**Config files that are empty**  
+these are just here to suppress errors
+* ```acl.conf```
+* ```ccss.conf```
+* ```cdr.conf```
+* ```cel.conf```
+* ```features.conf```
+* ```pjproject.conf```
+* ```pjsip_wizard.conf```
+
+
+
+**Config files for basic usage**  
+If a config file does not have a comment is does not require any tweaking  
+`*` Must be updated before first use
+* ```asterisk.conf```
+* ```logger.conf```
+* ```modules.conf``` 
+* ```udptl.conf```
+* ```chan_mobile.conf``` Settings for bluetooth `*`
+* ```extentions.conf``` Dial plan for extension
+* ```indications.conf``` Settings for dial tones
+* ```pjsip.conf``` Sets listening for PJSIP and defines the credentials for the extensions `*`

Linux/Config-Examples/Asterisk-BT_bridge/acl.conf

@@ -0,0 +1 @@
+; Asterisk ACL configuration

Linux/Config-Examples/Asterisk-BT_bridge/asterisk.conf

@@ -0,0 +1,19 @@
+; Asterisk core configuration
+[directories](!)
+astcachedir => /tmp
+astetcdir => /etc/asterisk
+astmoddir => /usr/lib/asterisk/modules
+astvarlibdir => /var/lib/asterisk
+astdbdir => /var/lib/asterisk
+astkeydir => /var/lib/asterisk
+astdatadir => /var/lib/asterisk
+astagidir => /var/lib/asterisk/agi-bin
+astspooldir => /var/spool/asterisk
+astrundir => /var/run/asterisk
+astlogdir => /var/log/asterisk
+astsbindir => /usr/sbin
+
+[options]
+;verbose = 3
+;debug = 3
+documentation_language = en_US

Linux/Config-Examples/Asterisk-BT_bridge/ccss.conf

@@ -0,0 +1 @@
+; Asterisk ccss (Call Completion Supplementary Services) configuration

Linux/Config-Examples/Asterisk-BT_bridge/cdr.conf

@@ -0,0 +1 @@
+; Asterisk cdr (Call Detail Record engine) configuration

Linux/Config-Examples/Asterisk-BT_bridge/cel.conf

@@ -0,0 +1 @@
+; Asterisk CEL (Channel Event Logging) configuration

Linux/Config-Examples/Asterisk-BT_bridge/chan_mobile.conf

@@ -0,0 +1,13 @@
+; Asterisk Chan Mobile configuration
+[general]
+interval=30
+
+[adapter]
+id=hci0
+address=<Adapter MAC> ;MAC of adapter
+
+[phone0]
+context=incoming-mobile
+address=<Phone MAC> ; MAC of Phone
+adapter=hci0   
+port=1   ;port number (from mobile search) 

Linux/Config-Examples/Asterisk-BT_bridge/extensions.conf

@@ -0,0 +1,12 @@
+; Asterisk exention configuration
+
+[from-internal]
+;Internal extentions (Phone,softphones or other Voip clients)
+exten => 7000,1,Dial(PJSIP/${EXTEN},30)
+
+;Outbound route
+exten => _X.,1,Dial(mobile/phone0/${EXTEN})
+
+;Inbound route
+[incoming-mobile]
+exten => s,1,Goto(from-internal,7000,1)

Linux/Config-Examples/Asterisk-BT_bridge/features.conf

@@ -0,0 +1 @@
+; Asterisk Call Features (transfer, monitor/mixmonitor, etc) configuration

Linux/Config-Examples/Asterisk-BT_bridge/indications.conf

@@ -0,0 +1,30 @@
+; Asterisk indication (location specific tone indications) configuration
+
+[general]
+country=nl
+
+[nl]
+description = Netherlands
+ringcadence = 1000,4000
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/4000
+congestion = 425/250,0/250
+callwaiting = 425/500,0/9500
+dialrecall = 425/500,0/50
+record = 1400/500,0/15000
+info = 950/330,1400/330,1800/330,0/1000
+stutter = 425/500,0/50
+
+[us]
+description = United States / North America
+ringcadence = 2000,4000
+dial = 350+440
+busy = 480+620/500,0/500
+ring = 440+480/2000,0/4000
+congestion = 480+620/250,0/250
+callwaiting = 440/300,0/10000
+dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+record = 1400/500,0/15000
+info = !950/330,!1400/330,!1800/330,0
+stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440

Linux/Config-Examples/Asterisk-BT_bridge/logger.conf

@@ -0,0 +1,7 @@
+; Asterisk logging configuration
+
+[general]
+
+[logfiles]
+console => notice,warning,error
+messages => notice,warning,error

Linux/Config-Examples/Asterisk-BT_bridge/modules.conf

@@ -0,0 +1,76 @@
+; Asterisk Module Loader configuration file
+
+[modules]
+autoload=no
+
+;Core
+load = app_dial.so
+load = pbx_config.so
+load = pbx_loopback.so
+load = pbx_realtime.so
+load = pbx_spool.so
+load = func_callerid.so
+load = func_dialplan.so
+load = func_dialgroup.so
+load = res_ael_share.so
+load = res_timing_timerfd.so
+
+;Codecs
+load = codec_alaw.so
+load = codec_g722.so
+load = codec_g726.so
+load = codec_gsm.so
+load = codec_opus_open_source.so
+load = codec_resample.so
+load = codec_ulaw.so
+
+;Formats
+load = format_g719.so
+load = format_g726.so
+load = format_gsm.so
+load = format_siren14.so
+load = format_siren7.so
+load = format_wav.so
+load = format_wav_gsm.so
+load = res_format_attr_g729.so
+load = res_format_attr_opus.so
+load = res_format_attr_siren14.so
+load = res_format_attr_siren7.so
+
+;PJSIP
+require = chan_pjsip.so
+load = func_pjsip_aor.so
+load = func_pjsip_endpoint.so
+load = res_pjproject.so
+load = res_pjsip.so
+load = res_pjsip_authenticator_digest.so
+load = res_pjsip_caller_id.so
+load = res_pjsip_endpoint_identifier_ip.so
+load = res_pjsip_endpoint_identifier_user.so
+load = res_pjsip_exten_state.so
+load = res_pjsip_outbound_publish.so
+load = res_pjsip_outbound_registration.so
+load = res_pjsip_path.so
+load = res_pjsip_pubsub.so
+load = res_pjsip_refer.so
+load = res_pjsip_registrar.so
+load = res_pjsip_sdp_rtp.so
+load = res_pjsip_session.so
+
+;RTP
+load = chan_rtp.so
+load = bridge_native_rtp.so
+load = res_rtp_asterisk.so
+load = res_rtp_multicast.so
+
+;Bridge
+load = bridge_simple.so
+load = bridge_builtin_features.so
+
+;Sorcery
+load = res_sorcery_astdb.so
+load = res_sorcery_config.so
+load = res_sorcery_memory.so
+
+;BT Interface
+require = chan_mobile.so

Linux/Config-Examples/Asterisk-BT_bridge/pjproject.conf

@@ -0,0 +1 @@
+; Asterisk pjproject options

Linux/Config-Examples/Asterisk-BT_bridge/pjsip.conf

@@ -0,0 +1,30 @@
+; Asterisk PJSIP config
+
+[transport-udp]
+type=transport
+protocol=udp
+bind=0.0.0.0
+
+;Internal VoipClient
+[7000]
+type=endpoint
+context=from-internal
+callerid=DeskPhone <7000>
+disallow=all
+;Some phone may reqire other codecs,
+;the following are available ulaw,alaw,gsm,g726,g722,opus
+allow=g722
+auth=7000
+aors=7000
+
+[7000]
+type=auth
+auth_type=userpass
+password=<Password>
+username=7000
+
+[7000]
+type=aor
+; if one device is connected setting '1' is okay
+; when a device sometimes fails to register (like my Friz!box) try increasing it, '5' is a safe value
+max_contacts=1

Linux/Config-Examples/Asterisk-BT_bridge/pjsip_wizard.conf

@@ -0,0 +1 @@
+; Asterisk PJSIP Wizard configuration

Linux/Config-Examples/Asterisk-BT_bridge/udptl.conf

@@ -0,0 +1,8 @@
+; Asterisk udptl config
+
+[general]
+udptlstart=4000
+udptlend=4999
+udptlfecentries = 3
+udptlfecspan = 3
+use_even_ports = no

Linux/Config-Examples/Asterisk-basic-server/ReadMe.md

@@ -0,0 +1,38 @@
+# Basic asterisk sever
+Go [Here](../../Docs/Setup-Asterisk-Server.md) to read about how to set it up
+
+
+**Config files that are empty**  
+these are just here to suppress errors
+* ```acl.conf```
+* ```ccss.conf```
+* ```cdr.conf```
+* ```cel.conf```
+* ```features.conf```
+* ```manager.conf``` 
+* ```pjproject.conf```
+* ```pjsip_wizard.conf```
+
+
+
+**Config files for basic usage**  
+If a config file does not have a comment, it does not require any tweaking  
+`*` Must be updated for each (new) extension  
+`**` Must be updated if (new) extension if a voip client needs to connect ({Soft}phone) or needs to be connected to (Homeassistant)
+* ```asterisk.conf```
+* ```extensions.conf``` Dial plan for extensions `*`
+* ```indications.conf``` Settings for dial tones
+* ```logger.conf```
+* ```modules.conf``` If not all functions are used you can disable some modules  
+* ```pjsip.conf``` Sets listening for PJSIP and loads sub configs
+* ```pjsip.extension.conf``` Defines the extension the credentials for the extensions `**`
+* ```pjsip.aor.conf``` Sets maximum connections for an extension `**`
+* ```pjsip.auth.conf``` Sets the credentials for the extensions`**`
+* ```udptl.conf```
+
+
+
+**Config files for use with ARI**  
+* ```ari.conf``` ARI basic config and credentials (Make sure to update this)
+* ```http.conf```  Settings for the integrated webserver for ARI
+* ```stasis.conf``` (Just here to suppress errors)

Linux/Config-Examples/Asterisk-basic-server/acl.conf

@@ -0,0 +1 @@
+; Asterisk ACL configuration

Linux/Config-Examples/Asterisk-basic-server/ari.conf

@@ -0,0 +1,9 @@
+; Asterisk ARI (Asterisk REST API) configuration
+[general]
+enabled = yes
+
+[aritest]
+type = user
+read_only = no
+password = testme
+password_format = plain

Linux/Config-Examples/Asterisk-basic-server/asterisk.conf

@@ -0,0 +1,19 @@
+; Asterisk core configuration
+[directories](!)
+astcachedir => /tmp
+astetcdir => /etc/asterisk
+astmoddir => /usr/lib/asterisk/modules
+astvarlibdir => /var/lib/asterisk
+astdbdir => /var/lib/asterisk
+astkeydir => /var/lib/asterisk
+astdatadir => /var/lib/asterisk
+astagidir => /var/lib/asterisk/agi-bin
+astspooldir => /var/spool/asterisk
+astrundir => /var/run/asterisk
+astlogdir => /var/log/asterisk
+astsbindir => /usr/sbin
+
+[options]
+;verbose = 3
+;debug = 3
+documentation_language = en_US

Linux/Config-Examples/Asterisk-basic-server/ccss.conf

@@ -0,0 +1 @@
+; Asterisk ccss (Call Completion Supplementary Services) configuration

Linux/Config-Examples/Asterisk-basic-server/cdr.conf

@@ -0,0 +1 @@
+; Asterisk cdr (Call Detail Record engine) configuration

Linux/Config-Examples/Asterisk-basic-server/cel.conf

@@ -0,0 +1 @@
+; Asterisk CEL (Channel Event Logging) configuration

Linux/Config-Examples/Asterisk-basic-server/extensions.conf

@@ -0,0 +1,45 @@
+; Asterisk exention configuration
+
+[from-internal]
+;Play audio message and hangup if 6010 is dialed
+exten = 6010,1,Answer()
+same = n,Wait(2)
+same = n,Playback(demo-congrats)
+same = n,Hangup()
+
+;Send http request, play audio message and hangup if 6011 is dialed
+exten = 6011,1,Answer()
+same = n,Wait(2)
+same = n,Set(CURL_RESULT=${CURL(http://<NoderedIP>:1880/endpoint/<ENDPOINT NAME>?exten=${EXTEN})})
+same = n,Playback(activated)
+same = n,Hangup()
+
+;Internal extentions (Phone,softphones or other Voip clients)
+; The number at the end (,30) sets the dial time out 
+exten => 6000,1,Dial(PJSIP/${EXTEN},30)
+
+;Connect to extenal the Homeassistent 'Assistent' service when 6020 is dialed
+exten => 6020,1,Dial(PJSIP/HomeAssistant,10)
+
+;IVR,
+;send to ivr if 6012 is dialed
+exten = 6012,1,Goto(ivr-test,s,1)
+
+;;The following section must be at the end of the file 
+[ivr-test]
+exten =>s,1,Answer()
+same =>n,set(IVRName=TestIVR)
+same => n,wait(2)
+same => n,Background(basic-pbx-ivr-main)
+same => n,WaitExten(10)
+;;Nr 1 to 4 are vaild options
+exten => _[1-4],1,Set(CURL_RESULT=${CURL(http://<NoderedIP>:1880/endpoint/<ENDPOINT NAME>?${IVRName}=${EXTEN})}
+same = n,Playback(activated)
+same = n,Playback(goodbye)
+same = n,Hangup()
+;;If option invalid
+exten => i,1,Playback(pm-invalid-option)
+exten => i,2,Goto(ivr-test,s,5)
+;; If timed out
+exten => t,1,Playback(vm-pls-try-again)
+exten => t,2,Goto(ivr-test,s,5)

Linux/Config-Examples/Asterisk-basic-server/features.conf

@@ -0,0 +1 @@
+; Asterisk Call Features (transfer, monitor/mixmonitor, etc) configuration

Linux/Config-Examples/Asterisk-basic-server/http.conf

@@ -0,0 +1,5 @@
+; Asterisk HTTP server (for ARI, etc) configuration
+[general]
+enabled=yes
+bindaddr=0.0.0.0
+bindport=8088

Linux/Config-Examples/Asterisk-basic-server/indications.conf

@@ -0,0 +1,30 @@
+; Asterisk indication (location specific tone indications) configuration
+
+[general]
+country=nl
+
+[nl]
+description = Netherlands
+ringcadence = 1000,4000
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/4000
+congestion = 425/250,0/250
+callwaiting = 425/500,0/9500
+dialrecall = 425/500,0/50
+record = 1400/500,0/15000
+info = 950/330,1400/330,1800/330,0/1000
+stutter = 425/500,0/50
+
+[us]
+description = United States / North America
+ringcadence = 2000,4000
+dial = 350+440
+busy = 480+620/500,0/500
+ring = 440+480/2000,0/4000
+congestion = 480+620/250,0/250
+callwaiting = 440/300,0/10000
+dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+record = 1400/500,0/15000
+info = !950/330,!1400/330,!1800/330,0
+stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440

Linux/Config-Examples/Asterisk-basic-server/logger.conf

@@ -0,0 +1,7 @@
+; Asterisk logging configuration
+
+[general]
+
+[logfiles]
+console => notice,warning,error
+messages => notice,warning,error

Linux/Config-Examples/Asterisk-basic-server/manager.conf

@@ -0,0 +1 @@
+; Asterisk AMI - (Asterisk Manager Interface) configuration

Linux/Config-Examples/Asterisk-basic-server/modules.conf

@@ -0,0 +1,109 @@
+; Asterisk Module Loader configuration file
+
+[modules]
+autoload=no
+
+;Core (BasicCalling)
+load = app_dial.so
+load = pbx_config.so
+load = pbx_loopback.so
+load = pbx_realtime.so
+load = pbx_spool.so
+load = func_callerid.so
+load = func_dialplan.so
+load = func_dialgroup.so
+load = res_ael_share.so
+load = res_timing_timerfd.so
+
+;Codecs (BasicCalling)
+load = codec_alaw.so
+load = codec_g722.so
+load = codec_g726.so
+load = codec_gsm.so
+load = codec_opus_open_source.so
+load = codec_resample.so
+load = codec_ulaw.so
+
+;Formats (BasicCalling)
+load = format_g719.so
+load = format_g726.so
+load = format_gsm.so
+load = format_ogg_speex.so
+load = format_siren14.so
+load = format_siren7.so
+load = format_wav.so
+load = format_wav_gsm.so
+load = res_format_attr_g729.so
+load = res_format_attr_opus.so
+load = res_format_attr_siren14.so
+load = res_format_attr_siren7.so
+
+;PJSIP (BasicCalling)
+require = chan_pjsip.so
+load = func_pjsip_aor.so
+load = func_pjsip_endpoint.so
+load = res_pjproject.so
+load = res_pjsip.so
+load = res_pjsip_authenticator_digest.so
+load = res_pjsip_caller_id.so
+load = res_pjsip_endpoint_identifier_ip.so
+load = res_pjsip_endpoint_identifier_user.so
+load = res_pjsip_exten_state.so
+load = res_pjsip_outbound_publish.so
+load = res_pjsip_outbound_registration.so
+load = res_pjsip_path.so
+load = res_pjsip_pubsub.so
+load = res_pjsip_refer.so
+load = res_pjsip_registrar.so
+load = res_pjsip_sdp_rtp.so
+load = res_pjsip_session.so
+
+;Sorcery (Rqired by PJSIP)
+load = res_sorcery_astdb.so
+load = res_sorcery_config.so
+load = res_sorcery_memory.so
+
+;RTP (Reqired by BasicCalling)
+load = chan_rtp.so
+load = bridge_native_rtp.so
+load = res_rtp_asterisk.so
+load = res_rtp_multicast.so
+
+;Bridge (Reqired by BasicCalling)
+load = bridge_simple.so
+load = bridge_builtin_features.so
+
+;FilePlayback (Reqired for playing a audio message for dialplan or IVR)
+load = app_controlplayback.so
+load = app_playback.so
+
+;;Make call from asterisk cli
+;load = app_originate.so
+;load = res_clioriginate.so 
+
+;ARI (Asterisk RESTful Interface)
+load => res_ari.so
+load => res_ari_applications.so
+load => res_ari_asterisk.so
+load => res_ari_bridges.so
+load => res_ari_channels.so
+load => res_ari_device_states.so
+load => res_ari_endpoints.so
+;load => res_ari_events.so
+load => res_ari_model.so
+load => res_ari_playbacks.so
+load => res_ari_recordings.so
+load => res_ari_sounds.so
+
+; stasis (Reqired by ARI)
+load => app_stasis.so
+load => res_stasis.so
+load => res_stasis_answer.so
+load => res_stasis_device_state.so
+load => res_stasis_playback.so
+load => res_stasis_recording.so
+load => res_stasis_snoop.so
+
+;cURL (Reqired for send curl commands form dialplan)
+load = func_curl.so
+load = res_curl.so

Linux/Config-Examples/Asterisk-basic-server/pjproject.conf

@@ -0,0 +1 @@
+; Asterisk pjproject options

Linux/Config-Examples/Asterisk-basic-server/pjsip.aor.conf

@@ -0,0 +1,15 @@
+; Asterisk AOR (address of record) config
+
+;Address for extenal the Homeassistent 'Assistent' service
+[HomeAssistant]
+type=aor
+qualify_frequency=0
+contact=sip:<HomeAssistant IP>:5060
+
+;Internal extentions (Phone,softphones or other Voip clients)
+[6000]
+type=aor
+
+; if one device is connected setting '1' is okay
+; when a device sometimes fails to register (like my Friz!box) try increasing it, '5' is a safe value 
+max_contacts=5

Linux/Config-Examples/Asterisk-basic-server/pjsip.auth.conf

@@ -0,0 +1,15 @@
+; Asterisk PJSIP auth/credentials config
+
+;Credentials for extenal the Homeassistent 'Assistent' service
+[HomeAssistant]
+type=auth
+auth_type=userpass
+password=
+username=HA
+
+;Internal extentions (Phone,softphones or other Voip clients)
+[6000]
+type=auth
+auth_type=userpass
+password=<PASSWORD>
+username=6000

Linux/Config-Examples/Asterisk-basic-server/pjsip.conf

@@ -0,0 +1,10 @@
+; Asterisk PJSIP config
+
+#include pjsip.extension.conf
+#include pjsip.auth.conf
+#include pjsip.aor.conf
+
+[transport-udp]
+type=transport
+protocol=udp
+bind=0.0.0.0

Linux/Config-Examples/Asterisk-basic-server/pjsip.extension.conf

@@ -0,0 +1,32 @@
+; Asterisk PJSIP  extention config
+
+;settings Homeassistent 'Assistent' service extention
+[HomeAssistant]
+type=endpoint
+context=from-external
+disallow=all
+allow=opus
+aors=HA
+send_connected_line=false
+rtp_keepalive=0
+user_eq_phone=no
+trust_id_inbound=no
+direct_media=no
+rtp_symmetric=yes
+auth=HomeAssistant
+aors=HomeAssistant
+;dtmf_mode=auto
+
+;Internal extentions (Phone,softphones or other Voip clients)
+
+[6000]
+type=endpoint
+context=from-internal
+callerid=DeskPhone <6000>
+disallow=all
+;Some phone may reqire other codecs,
+;the following are available ulaw,alaw,gsm,g726,g722,opus
+allow=g722
+auth=6000
+aors=6000
+

Linux/Config-Examples/Asterisk-basic-server/pjsip_wizard.conf

@@ -0,0 +1 @@
+; Asterisk PJSIP Wizard configuration

Linux/Config-Examples/Asterisk-basic-server/stasis.conf

@@ -0,0 +1,4 @@
+; Asterisk Stasis config
+[threadpool]
+
+[declined_message_types]

Linux/Config-Examples/Asterisk-basic-server/udptl.conf

@@ -0,0 +1,8 @@
+; Asterisk udptl config
+
+[general]
+udptlstart=4000
+udptlend=4999
+udptlfecentries = 3
+udptlfecspan = 3
+use_even_ports = no

Linux/Docs/Setup-Asterisk-Server.md

@@ -0,0 +1,75 @@
+# Asterisk basic sever
+
+Features
+* Calling other connected/internal phones
+* Extension that plays an audio message
+* Extension that sends a http request (Node-Red for example) and plays an audio message
+* Extension that forwards the call the the Homeassistent 'Assistent' 
+* Extension with a IVR menu that will send the chosen option as a http request (Node-Red for example) and  plays an audio message
+
+
+## Installing Packages
+**Alpine Linux**
+
+```
+apk add asterisk asterisk-curl asterisk-opus asterisk-sounds-en asterisk-sounds-moh asterisk-speex asterisk-tds
+```
+
+## Configuring
+(Re)move the default asterisk config (`/etc/asterisk`).  
+Download the config files from here [Here](../Config-Examples/Asterisk-basic-server) and put them in the config directory `/etc/asterisk`.  
+Set the permissions `chown -R asterisk: /etc/asterisk`.  
+Edit the configuration files to suit your needs,  
+make sure to change the credentials, and check the module list 
+
+**Alpine Linux**
+```
+chown -R asterisk: /etc/asterisk
+rc-update add asterisk
+service asterisk start
+```
+
+# Asterisk bluetooth bridge
+[Source](https://hotkey404.com/connect-your-phone-to-raspbx-via-bluetooth-with-chan_mobile/)  
+Features
+* Bridge a mobile phone to a voip phone/pbx
+
+
+## Installing Packages
+**Alpine Linux**
+
+```
+apk add asterisk asterisk-opus asterisk-mobile udev bluez hidapi
+```
+
+## Configuring
+(Re)move the default asterisk config (`/etc/asterisk`).  
+Download the config files from here [Here](../Config-Examples/Asterisk-BT_bridge) and put them in the config directory `/etc/asterisk`.  
+Set the permissions `chown -R asterisk: /etc/asterisk`.  
+Edit the credentials for your (soft)phone in `/etc/asterisk/pjsip.conf`  
+Pair the phone using `bluetoothctl`
+Edit the phone and adapter bluethooth mac address  in `/etc/asterisk/chan_mobile.conf`  
+(Re)start Asterisk 
+Connect to Asterisk CLI `asterisk -r`  
+Make sure the phone is discoverable/in pairing mode  
+In Asterisk CLI run `mobile search`  
+Change the port to the port from mobile scan in `/etc/asterisk/chan_mobile.conf`  
+Restart Asterisk 
+
+**Alpine Linux**
+```
+chown -R asterisk: /etc/asterisk
+modprobe btusb
+modprobe uhid
+rc-update add asterisk
+rc-update add bluetooth
+service asterisk start
+```  
+
+**bluetoothctl**
+```
+#Open the bluetoothCLI
+bluetoothctl
+scan on
+pair <Mac Of phone>
+```

NodeRed/Readme.md

@@ -0,0 +1,4 @@
+# NodeRed Snippits
+## List of Flows
+## List of SubFlows
+* **ARI_caller** Uses the Asterisk REST API to generate a call between two extensions

NodeRed/SubFlow-ARI_Caller.json

@@ -0,0 +1,183 @@
+[
+  {
+    "id": "debed60bf171318a",
+    "type": "subflow",
+    "name": "ARI Caller",
+    "info": "`EndPoint`  \r\nPhone/Extention that rings\r\n\r\n`Extention`  \r\nPhone/Extention that gets connected after answer  \r\nMost likely it will play a message ",
+    "category": "",
+    "in": [{ "x": 40, "y": 40, "wires": [{ "id": "0c497bbfbb3e3e0e" }] }],
+    "out": [
+      { "x": 520, "y": 40, "wires": [{ "id": "0f9bb285c2559f1c", "port": 0 }] }
+    ],
+    "env": [
+      {
+        "name": "",
+        "type": "str",
+        "value": "Spacer",
+        "ui": { "type": "none" }
+      },
+      {
+        "name": "",
+        "type": "str",
+        "value": "ServerSettings",
+        "ui": {
+          "icon": "font-awesome/fa-server",
+          "label": { "en-US": "ARI Server" },
+          "type": "none"
+        }
+      },
+      {
+        "name": "ariServer",
+        "type": "str",
+        "value": "192.168.2.20",
+        "ui": {
+          "label": { "en-US": "IP" },
+          "type": "input",
+          "opts": { "types": ["str"] }
+        }
+      },
+      {
+        "name": "ariPort",
+        "type": "num",
+        "value": "8088",
+        "ui": {
+          "label": { "en-US": "Port" },
+          "type": "input",
+          "opts": { "types": ["num"] }
+        }
+      },
+      {
+        "name": "",
+        "type": "str",
+        "value": "Spacer",
+        "ui": { "type": "none" }
+      },
+      {
+        "name": "",
+        "type": "str",
+        "value": "ServerCredentials",
+        "ui": {
+          "icon": "font-awesome/fa-user",
+          "label": { "en-US": "ARI Credentials" },
+          "type": "none"
+        }
+      },
+      {
+        "name": "ariUser",
+        "type": "str",
+        "value": "",
+        "ui": {
+          "label": { "en-US": "ARI User" },
+          "type": "input",
+          "opts": { "types": ["str"] }
+        }
+      },
+      {
+        "name": "ariPassword",
+        "type": "cred",
+        "ui": {
+          "label": { "en-US": "ARI Password" },
+          "type": "input",
+          "opts": { "types": ["cred"] }
+        }
+      },
+      {
+        "name": "",
+        "type": "str",
+        "value": "Spacer",
+        "ui": { "type": "none" }
+      },
+      {
+        "name": "",
+        "type": "str",
+        "value": "CallSettings",
+        "ui": {
+          "icon": "font-awesome/fa-phone",
+          "label": { "en-US": "Call Setting" },
+          "type": "none"
+        }
+      },
+      {
+        "name": "endpoint",
+        "type": "num",
+        "value": "",
+        "ui": {
+          "label": { "en-US": "EndPoint" },
+          "type": "input",
+          "opts": { "types": ["num"] }
+        }
+      },
+      {
+        "name": "extention",
+        "type": "num",
+        "value": "",
+        "ui": {
+          "label": { "en-US": "Extention" },
+          "type": "input",
+          "opts": { "types": ["num"] }
+        }
+      },
+      {
+        "name": "CallerID",
+        "type": "str",
+        "value": "NodeRed <0>",
+        "ui": { "label": { "en-US": "Caller ID" } }
+      }
+    ],
+    "meta": { "author": "brammp <dev@bprieshof.nl>", "desc": "ARI Caller" },
+    "color": "#3FADB5",
+    "icon": "font-awesome/fa-phone"
+  },
+  {
+    "id": "0f9bb285c2559f1c",
+    "type": "http request",
+    "z": "debed60bf171318a",
+    "name": "Send ARI request",
+    "method": "POST",
+    "ret": "txt",
+    "paytoqs": "ignore",
+    "url": "",
+    "tls": "",
+    "persist": false,
+    "proxy": "",
+    "insecureHTTPParser": false,
+    "authType": "basic",
+    "senderr": false,
+    "headers": [
+      {
+        "keyType": "other",
+        "keyValue": "Content-Type",
+        "valueType": "other",
+        "valueValue": "application/x-www-form-urlencoded"
+      }
+    ],
+    "x": 390,
+    "y": 40,
+    "wires": [[]]
+  },
+  {
+    "id": "0c497bbfbb3e3e0e",
+    "type": "function",
+    "z": "debed60bf171318a",
+    "name": "Prepair ARI request",
+    "func": "var call={\n    url: env.get(\"ariServer\") + \":\" + env.get(\"ariPort\") + \"/ari/channels\",\n    payload:{ \n        'endpoint': 'PJSIP/' + env.get(\"endpoint\"),\n        'extension': env.get(\"extention\"),\n        'context': 'from-internal',\n        'callerId': env.get(\"CallerID\")\n    }\n\n};\nreturn call;\n\n\n\n\n",
+    "outputs": 1,
+    "timeout": 0,
+    "noerr": 0,
+    "initialize": "",
+    "finalize": "",
+    "libs": [],
+    "x": 170,
+    "y": 40,
+    "wires": [["0f9bb285c2559f1c"]]
+  },
+  {
+    "id": "603f44a955e26689",
+    "type": "subflow:debed60bf171318a",
+    "z": "5b5de3f9b5eae1d7",
+    "name": "",
+    "x": 80,
+    "y": 60,
+    "wires": [[]]
+  }
+]